[asterisk-users] Problem receiving audio with Asterisk 1.4.5 with SIP trunk

[Zaheer K. Master]

Hi All,

I'm running asterisk 1.4.5 (on AsteriskNOW beta 6 appliance) and using Snom
360/370 phones with direct SIP trunking from bandwidth.com. I can make
outgoing calls, and the person on the receiving end can hear my voice, but I
cannot hear them. I also cannot receive incoming calls to my DID number. 

Here is my current setup:
Asterisk is running on a dell poweredge server with an ip of 192.168.1.55
I have setup a 1:1 NAT for asterisk with my public IP of 72.127.218.XXX
The phones have IPs of 192.168.1.150-160
I can register my phones and they work correctly for intercom, voicemail,
etc.

I think the problem is that after the SIP session has initiated, the phones
are giving an IP of 192.168.1.151 for the return audio, and those packets
are getting dropped. I'm not sure where to go from here to get the incoming
calls/audio working. Do I have to give the Asterisk box a public IP? I tried
this, and when I did I was unable to get the phones to register - probably
since they had private IPs.

Any help or suggestions would be greatly appreciated. Thanks in advance!

Regards,
Zaheer